April 2020 - Alsa-devel - mailman.alsa-project.org

State of hda-analyser.py
by Moosa Mahsoom 18 Apr '20

18 Apr '20

Hi, I am a final year student from BITS Pilani, India. I have been facing some issues with my laptop audio and hence wanted to use the hda_analyser script. However, I noticed that the python script doesn't seem to compatible with the current pygtk libraries. If possible,I would like to contribute towards bringing the scripts to parity. Yours Sincerely, Moosa Mahsoom

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Bug WD19 dockstation
by Gregorio Rodrigo 17 Apr '20

17 Apr '20

Hi everyone, I am not sure if this is the proper way to submit a bug/ticket/question. I posted a message here https://github.com/edrose/dell-dock-audio-fix/issues/2#issuecomment-6152907… and got the suggestion to write to the alsa mail list. I have a WD19 docking station and the lineout is not recognized. It seems it is recognized as an input. However it works if I run the speaker-test command. Anyone knows if I can apply those UCM settings or how could I do it? I copy here the message. I have a WD19 docking station and i am facing the same problem. Debian GNU/Linux bullseye/sid \n \l Linux xps 5.5.0-1-amd64 #1 <https://github.com/edrose/dell-dock-audio-fix/issues/1> SMP Debian 5.5.13-2 (2020-03-30) x86_64 GNU/Linux pactl list cards Card #1 <https://github.com/edrose/dell-dock-audio-fix/issues/1> Name: alsa_card.usb-Generic_USB_Audio_200901010001-00 Driver: module-alsa-card.c Owner Module: 25 Properties: alsa.card = "2" alsa.card_name = "WD19 Dock" alsa.long_card_name = "Dell-WD15-Dock" alsa.driver_name = "snd_usb_audio" device.bus_path = "pci-0000:00:14.0-usb-0:9.3.4:1.0" sysfs.path = "/devices/pci0000:00/0000:00:14.0/usb3/3-9/3-9.3/3-9.3.4/3-9.3.4:1.0/sound/card2" udev.id = "usb-Generic_USB_Audio_200901010001-00" device.bus = "usb" device.vendor.id = "0bda" device.vendor.name = "Realtek Semiconductor Corp." device.product.id = "402e" device.product.name = "USB Audio" device.serial = "Generic_USB_Audio_200901010001" device.string = "2" device.description = "USB Audio" module-udev-detect.discovered = "1" device.icon_name = "audio-card-usb" Profiles: input:analog-stereo: Analog Stereo Input (sinks: 0, sources: 1, priority: 65, available: yes) output:analog-stereo: Analog Stereo Output (sinks: 1, sources: 0, priority: 6500, available: yes) output:analog-stereo+input:analog-stereo: Analog Stereo Duplex (sinks: 1, sources: 1, priority: 6565, available: yes) off: Off (sinks: 0, sources: 0, priority: 0, available: yes) Active Profile: output:analog-stereo+input:analog-stereo Ports: analog-input-mic: Microphone (priority: 8700, latency offset: 0 usec) Properties: device.icon_name = "audio-input-microphone" Part of profile(s): input:analog-stereo, output:analog-stereo+input:analog-stereo analog-output-headphones: Headphones (priority: 9900, latency offset: 0 usec) Properties: device.icon_name = "audio-headphones" Part of profile(s): output:analog-stereo, output:analog-stereo+input:analog-stereo http://alsa-project.org/db/?f=616054544b8748431c354f1affbfd705c88fad37 If i run manualy speaker-test and select the proper subdevice, i get sound over the lineoutput. I cannot set it as default (or i dont know how to do it). It seems that if i am able to set the UCM2 configuration from WD15 should work. Any advise? Thank you in advance.

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[PATCH] ASoC: stm32: sai: fix sai probe
by Olivier Moysan 17 Apr '20

17 Apr '20

pcm config must be set before snd_dmaengine_pcm_register() call. Fixes: 0d6defc7e0e4 ("ASoC: stm32: sai: manage rebind issue") Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai_sub.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 34a7c3d6fb91..41f01c3e639e 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1547,6 +1547,9 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) return ret; } + if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) + conf = &stm32_sai_pcm_config_spdif; + ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0); if (ret) { if (ret != -EPROBE_DEFER) @@ -1556,15 +1559,10 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) ret = snd_soc_register_component(&pdev->dev, &stm32_component, &sai->cpu_dai_drv, 1); - if (ret) { + if (ret) snd_dmaengine_pcm_unregister(&pdev->dev); - return ret; - } - - if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) - conf = &stm32_sai_pcm_config_spdif; - return 0; + return ret; } static int stm32_sai_sub_remove(struct platform_device *pdev) -- 2.17.1

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[PATCH] ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/Broadwell
by Pierre-Louis Bossart 17 Apr '20

17 Apr '20

Major regressions were detected by SOF CI on CherryTrail and Broadwell: [ 25.705750] SSP2-Codec: ASoC: no backend playback stream [ 27.923378] SSP2-Codec: ASoC: no users playback at close - state This is root-caused to the introduction of the DAI capability checks with snd_soc_dai_stream_valid(). Its use in soc-pcm.c makes it a requirement for all DAIs to report at least a non-zero min_channels field. For some reason the SSP structures used for SKL+ did provide this information but legacy platforms didn't. Fixes: 9b5db059366ae2 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported") Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/bdw.c | 16 +++++++++++++ sound/soc/sof/intel/byt.c | 48 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 64 insertions(+) diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c index 6c23c5769330..a32a3ef78ec5 100644 --- a/sound/soc/sof/intel/bdw.c +++ b/sound/soc/sof/intel/bdw.c @@ -567,9 +567,25 @@ static void bdw_set_mach_params(const struct snd_soc_acpi_mach *mach, static struct snd_soc_dai_driver bdw_dai[] = { { .name = "ssp0-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp1-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, }; diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index f84391294f12..29fd1d86156c 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -459,21 +459,69 @@ static void byt_set_mach_params(const struct snd_soc_acpi_mach *mach, static struct snd_soc_dai_driver byt_dai[] = { { .name = "ssp0-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp1-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp2-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + } }, { .name = "ssp3-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp4-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, { .name = "ssp5-port", + .playback = { + .channels_min = 1, + .channels_max = 8, + }, + .capture = { + .channels_min = 1, + .channels_max = 8, + }, }, }; -- 2.20.1

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[PATCH 0/2] ASoC: rsnd: Fixes for multichannel HDMI audio output
by Matthias Blankertz 17 Apr '20

17 Apr '20

This fixes two issues in the snd-soc-rcar driver blocking multichannel HDMI audio out: The parent SSI in a multi-SSI configuration is not correctly set up and started, and the SSI->HDMI channel mapping is wrong. With these patches, the following device tree snippet can be used on an r8a7795-based platform (Salvator-X) to enable multichannel HDMI audio on HDMI0: rsnd_port1: port@1 { rsnd_endpoint1: endpoint { remote-endpoint = <&dw_hdmi0_snd_in>; dai-format = "i2s"; bitclock-master = <&rsnd_endpoint1>; frame-master = <&rsnd_endpoint1>; playback = <&ssi0 &ssi1 &ssi2 &ssi9>; }; }; With a capable receiver attached, all of 2ch (stereo), 6ch (e.g. 5.1) and 8ch audio output should work. Matthias Blankertz (2): ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode ASoC: rsnd: Fix HDMI channel mapping for multi-SSI mode sound/soc/sh/rcar/ssi.c | 8 ++++---- sound/soc/sh/rcar/ssiu.c | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) base-commit: 7111951b8d4973bda27ff663f2cf18b663d15b48 -- 2.26.0

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Question about snd_soc_card_register()
by Sridharan, Ranjani 17 Apr '20

17 Apr '20

Hi Takashi, While working on implementing the probes features in SOF using a separate card for the probe DAI links, I noticed that calling snd_soc_register_card() results in incrementing the usage_count for the device that registers the card by 2 and it is not decremented until the card is freed. Is this the expected behaviour? Typically, we register a separate platform device for the Intel machines which in turn register the card and none of them ever enable runtime PM. So this has no impact on the parent device's runtime PM status. I'd like to avoid creating a separate platform device just to register the card if possible while also enabling runtime PM . But when I do this today, the device cannot enter runtime suspend at all. Could you please shed some light on this? Thanks, Ranjani

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Re: next/master bisection: baseline.dmesg.alert on meson-sm1-sei610
by Mark Brown 17 Apr '20

17 Apr '20

On Fri, Apr 17, 2020 at 02:53:36AM -0700, kernelci.org bot wrote: > next/master bisection: baseline.dmesg.alert on meson-sm1-sei610 > Summary: > Start: a3ca59b9af21e Add linux-next specific files for 20200416 > Plain log: https://storage.kernelci.org//next/master/next-20200416/arm64/defconfig+CON… > HTML log: https://storage.kernelci.org//next/master/next-20200416/arm64/defconfig+CON… > Result: 9b5db059366ae ASoC: soc-pcm: dpcm: Only allow playback/capture if supported This change to check the playback and capture constraints is also causing issues on at least Meson - I'm a bit worried that this is also causing oopses here, not just audio problems so copying in Morimoto-san too. We should fix the things that are broken by this change since it's a thing we should be able to rely on and do enforce for non-DPCm links but I'm wondering if we should revert for 5.7 and reapply for 5.8. Including complete report so people have it. > Checks: > revert: PASS > verify: PASS > Parameters: > Tree: next > URL: https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git > Branch: master > Target: meson-sm1-sei610 > CPU arch: arm64 > Lab: lab-baylibre > Compiler: gcc-8 > Config: defconfig+CONFIG_RANDOMIZE_BASE=y > Test case: baseline.dmesg.alert > > Breaking commit found: > > ------------------------------------------------------------------------------- > commit 9b5db059366ae2087e07892b5fc108f81f4ec189 > Author: Stephan Gerhold <stephan(a)gerhold.net> > Date: Wed Apr 15 12:49:28 2020 +0200 > > ASoC: soc-pcm: dpcm: Only allow playback/capture if supported > > At the moment, PCM devices for DPCM are only created based on the > dpcm_playback/capture parameters of the DAI link, without considering > if the CPU/FE DAI is actually capable of playback/capture. > > Normally the dpcm_playback/capture parameter should match the > capabilities of the CPU DAI. However, there is no way to set that > parameter from the device tree (e.g. with simple-audio-card or > qcom sound cards). dpcm_playback/capture are always both set to 1. > > This causes problems when the CPU DAI does only support playback > or capture. Attemting to open that PCM device with an unsupported > stream type then results in a null pointer dereference: > > Unable to handle kernel NULL pointer dereference at virtual address 0000000000000128 > Internal error: Oops: 96000044 [#1] PREEMPT SMP > CPU: 3 PID: 1582 Comm: arecord Not tainted 5.7.0-rc1 > pc : invalidate_paths_ep+0x30/0xe0 > lr : snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 > Call trace: > invalidate_paths_ep+0x30/0xe0 > snd_soc_dapm_dai_get_connected_widgets+0x170/0x1a8 > dpcm_path_get+0x38/0xd0 > dpcm_fe_dai_open+0x70/0x920 > snd_pcm_open_substream+0x564/0x840 > snd_pcm_open+0xfc/0x228 > snd_pcm_capture_open+0x4c/0x78 > snd_open+0xac/0x1a8 > ... > > ... because the DAI playback/capture_widget is not set in that case. > > We could add checks there to fix the problem (maybe we should > anyway), but much easier is to not expose the device as > playback/capture in the first place. Attemting to use that > device would always fail later anyway. > > Add checks for snd_soc_dai_stream_valid() to the DPCM case > to avoid exposing playback/capture if it is not supported. > > Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> > Link: https://lore.kernel.org/r/20200415104928.86091-1-stephan@gerhold.net > Signed-off-by: Mark Brown <broonie(a)kernel.org> > > diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c > index 289aebc155293..1f302de440525 100644 > --- a/sound/soc/soc-pcm.c > +++ b/sound/soc/soc-pcm.c > @@ -2911,8 +2911,17 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) > int i; > > if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { > - playback = rtd->dai_link->dpcm_playback; > - capture = rtd->dai_link->dpcm_capture; > + cpu_dai = asoc_rtd_to_cpu(rtd, 0); > + if (rtd->num_cpus > 1) { > + dev_err(rtd->dev, > + "DPCM doesn't support Multi CPU yet\n"); > + return -EINVAL; > + } > + > + playback = rtd->dai_link->dpcm_playback && > + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK); > + capture = rtd->dai_link->dpcm_capture && > + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE); > } else { > /* Adapt stream for codec2codec links */ > int cpu_capture = rtd->dai_link->params ? > ------------------------------------------------------------------------------- > > > Git bisection log: > > ------------------------------------------------------------------------------- > git bisect start > # good: [87b0f983f66f23762921129fd35966eddc3f2dae] net: mscc: ocelot: fix untagged packet drops when enslaving to vlan aware bridge > git bisect good 87b0f983f66f23762921129fd35966eddc3f2dae > # bad: [a3ca59b9af21e68069555ffff1ad89bd2a7c40fc] Add linux-next specific files for 20200416 > git bisect bad a3ca59b9af21e68069555ffff1ad89bd2a7c40fc > # bad: [feb09551bfe34ccf0ba462188a1aee651be0f2c3] Merge remote-tracking branch 'i2c/i2c/for-next' > git bisect bad feb09551bfe34ccf0ba462188a1aee651be0f2c3 > # good: [a4721ced760684d1776bf31f7925aa41bb3f4846] Merge v5.7-rc1 into drm-misc-fixes > git bisect good a4721ced760684d1776bf31f7925aa41bb3f4846 > # bad: [e75043c693af6a10c9e2087adeef243cf05ce3bd] Merge remote-tracking branch 'rockchip/for-next' > git bisect bad e75043c693af6a10c9e2087adeef243cf05ce3bd > # bad: [08a315860a1b462fef5969d2e6ab4cdab26df8e3] Merge remote-tracking branch 'arm-soc/for-next' > git bisect bad 08a315860a1b462fef5969d2e6ab4cdab26df8e3 > # bad: [80dbfa57508c36c2b3798b387ef3eb92954b9c09] Merge remote-tracking branch 'slave-dma-fixes/fixes' > git bisect bad 80dbfa57508c36c2b3798b387ef3eb92954b9c09 > # good: [a723380c6f0a3723db72da6c6097b32b23f5564c] Merge remote-tracking branch 'sound-current/for-linus' > git bisect good a723380c6f0a3723db72da6c6097b32b23f5564c > # bad: [5bd70440cb0a6f5c6a84019bb2aa93ab8310a5cd] ASoC: soc-dai: revert all changes to DAI startup/shutdown sequence > git bisect bad 5bd70440cb0a6f5c6a84019bb2aa93ab8310a5cd > # good: [4d1a015a203c0249e3332ea217a38ec978118daa] ASoC: convert rockchip spdif bindings to yaml > git bisect good 4d1a015a203c0249e3332ea217a38ec978118daa > # good: [0f2a3b02274c02eb97697c4d89c019d1d21ac225] ASoC: wsa881x: mark read_only_wordlength flag > git bisect good 0f2a3b02274c02eb97697c4d89c019d1d21ac225 > # good: [aa7812737f2877e192d57626cbe8825cc7cf6de9] ASoC: sgtl5000: Fix VAG power-on handling > git bisect good aa7812737f2877e192d57626cbe8825cc7cf6de9 > # bad: [0c824ec094b5cda766c80d88c2036e28c24a4cb1] ASoC: q6dsp6: q6afe-dai: add missing channels to MI2S DAIs > git bisect bad 0c824ec094b5cda766c80d88c2036e28c24a4cb1 > # bad: [9b5db059366ae2087e07892b5fc108f81f4ec189] ASoC: soc-pcm: dpcm: Only allow playback/capture if supported > git bisect bad 9b5db059366ae2087e07892b5fc108f81f4ec189 > # first bad commit: [9b5db059366ae2087e07892b5fc108f81f4ec189] ASoC: soc-pcm: dpcm: Only allow playback/capture if supported > -------------------------------------------------------------------------------

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[PATCH] ASoC: Intel: atom: remove unused 'sst_mix_voip_controls'
by Jason Yan 17 Apr '20

17 Apr '20

Fix the following gcc warning: sound/soc/intel/atom/sst-atom-controls.c:710:30: warning: ‘sst_mix_voip_controls’ defined but not used [-Wunused-const-variable=] SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_voip_controls); ^ Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Jason Yan <yanaijie(a)huawei.com> --- sound/soc/intel/atom/sst-atom-controls.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index 69f3af4524ab..541f0059ba7a 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -707,7 +707,6 @@ SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_pcm2_controls); SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_sprot_l0_controls); SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_media_l1_controls); SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_media_l2_controls); -SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_voip_controls); SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_codec0_controls); SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_codec1_controls); SST_SBA_DECLARE_MIX_CONTROLS(sst_mix_modem_controls); -- 2.21.1

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[PATCH v2] ASoC: bdw-rt5650: remove 3-channel capture support
by Brent Lu 17 Apr '20

17 Apr '20

Implement a constrain to exclude 3-channel capture since only 2 and 4 channel capture are supported on the platform. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/bdw-rt5650.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c index af2f502..eedbdad 100644 --- a/sound/soc/intel/boards/bdw-rt5650.c +++ b/sound/soc/intel/boards/bdw-rt5650.c @@ -83,6 +83,36 @@ static struct snd_soc_jack_pin mic_jack_pin = { .mask = SND_JACK_MICROPHONE, }; +static const unsigned int channels[] = { + 2, 4, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static int bdw_fe_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + /* + * On this platform for PCM device we support, + * 2 or 4 channel capture + */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + return 0; +} + +static const struct snd_soc_ops bdw_rt5650_fe_ops = { + .startup = bdw_fe_startup, +}; + static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -234,6 +264,7 @@ static struct snd_soc_dai_link bdw_rt5650_dais[] = { .name = "System PCM", .stream_name = "System Playback", .dynamic = 1, + .ops = &bdw_rt5650_fe_ops, #if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) .init = bdw_rt5650_rtd_init, #endif -- 2.7.4

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[PATCH alsa-lib] conf: USB-Audio: Fix S/PDIF output of ASUS Xonar AE
by Ivan Mironov 17 Apr '20

17 Apr '20

ASUS Xonar AE is a PCI-Express card containing USB controller: USB controller [0c03]: ASMedia Technology Inc. ASM1042A USB 3.0 Host Controller [1b21:1142] and the actual USB sound card: ID 0b05:180f ASUSTek Computer, Inc. XONAR SOUND CARD As other Xonar USB sound cards, it uses second device for digital output. Signed-off-by: Ivan Mironov <mironov.ivan(a)gmail.com> diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 89811086db1d..3f4c421b7e2c 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -40,6 +40,7 @@ USB-Audio.pcm.iec958_device { "USB Sound Blaster HD" 1 "Xonar U7" 1 "ASUS XONAR U5" 1 + "XONAR SOUND CARD" 1 # The below don't have digital in/out, so prevent them from being opened. "Andrea PureAudio USB-SA Headset" 999 -- 2.25.1

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